The "reference" is just the original convolved with an impulse, i.e. a null operation.

All files are 24-bit 96kHz sampled .wav files. You will need a 24/96 sound card to listen to these files, and use something like foobar2k which plays the files properly.

The "reference" should sound best, the "maximum phase" file should sound worst. Can anyone ABX a difference?

(I don't claim to be able to hear any difference).

btw, the files are time aligned to the nearest sample, though the maximum phase filter introduces a sub-sample delay and slight phase shift. I've confirmed that all low pass filters have identical amplitude responses, and that the pass band of the low pass filters matches the level of the unit impulse. Source: Cool Edit Pro frequency analysis, 65536 FFT, Linear View, Range 24 dB, Reference -90 dBFS, window expanded to fill my screen, absolutely no different between plots.

In the last second end of each audio file is a unit impulse, filtered - hence the filter itself is "included" in the file, for easy checking, analysis and verification.

(I used Cool Edit Pro to do the convolution (filtering). I'm not entirely convinced by its algorithm - there are some errors 100dB below the signal, verified by convolving with a single impulse. From the nature of the errors, I assume it's performing time domain convolution by frequency domain multiplication, which is quite common, and there are some rounding errors which would be lost in the noise of a 16-bit signal, but are apparent with the 24-bit signal)

Cheers,David.

P.S. When I tried to FLAC encode these files (using FLAC 1.2.1b in Windows XP), the FLAC verifier from the same package (not the one called automatically when encoding, but the stand-alone one) reported errors, hence I've provided .wavs instead. I hope my system isn't on the way out - can anyone else reproduce this problem?

I've attached some spectral plots to show how significant the ringing is at 20kHz.

limehouse1.jpg shows the original (top) and filtered (bottom).

The broad shaded vertical lines are drum hits, the little horizontal flecks at the top left of them in the bottom image are the pre-ringing of the maximum phase filter.

The vertical dotted line represents the cursor. limehouse2.jpg shows the spectrum at the cursor. You can see the spike at 20kHz. It's about 10dB less than the energy in the drum hit itself (not shown).

I, probably like most people here, can't hear a whit above 18k to begin with. So I'm not even going to try.

According to people who believe in such things, that shouldn't disqualify you at all...

QUOTE

The preference for higher sample rates is not based on ultrasonic content, but on time resolution and aliasing when conversion goes wrong. All anti-alias filters are brick wall, anti-causal (i.e. pre-ring) which is unnatural. You can hear the difference between different sample rates, even when the tweeter doesn't go above 18kHz and/or your hearing stops at 17kHz.

David,I tried the reference against the maximum phase version. I decided on just 5 trials, as I find ABX listening quite a strain, but 5 trials gives a fairly convincing outcome if all answers are correct. I find a segment of about 1 or two seconds' duration that seems to sound different. I then try to avoid answering until I am quite sure of the answer.

I took a long refresh break for trials 4 and 5 because my hearing had temporarily lost its ability to discriminate fine differences. Also the segment I chose to listen to for trials 4 and 5 was different from the segment I used for trials 1, 2 and 3.

I do not have particularly good high frequency hearing. These days it cuts out before 20KHz.

It is of course always possible that something in my playback equipment created the difference. There are always some lingering doubts about results like these.

My playback system is a home theatre pc with an onboard high definition sound chip outputting discrete analogue channels. The analogue channels are connected to an AVR, driving medium quality hi-fi speakers.

To make sure the files remain accessible, I've also put them online on my website.Note that these are slightly modified files made on Jul 9 2010 by bandpass.The shell-script used to generate the latest samples is available as well.

The maximum phase version sounds not so pristine, transients seemed to have less "bite" (the clapping for instance). And I thought that the room response changed. I always write a sentence like this one: the room holding the microphones together vanishes. Well, sort of.

Independently from each other I thought that both versions sounded very good. What a marvelously engineered recording!

That's two independent positive ABX results - now this is getting interesting!

QUOTE (Cavaille @ Jul 10 2010, 02:41)

Independently from each other I thought that both versions sounded very good. What a marvelously engineered recording!

It certainly sounds like you're in the room - though I'm not 100% happy with the sound that's in the room.

I think this recording is astonishing...http://www.soundkeeperrecordings.com/format.htm...though I don't believe it's as suitable for revealing the potential effects of a 20k LPF so well. I could be completely wrong of course! I'll have a play next week if I remember.

Unless it is on my end, the rapidshare download page won't load. I only gave it a go, not thinking I'll have any luck at differentiating the filters, only because I was curious what the music and or signal would be and there was an invitation to give it a go in another thread today.

If it's a hassle to reinstate the files, don't bother on my behalf. It's not a biggie, I just thought I'd point it out.

As discussed in another thread it seems that foo_abx has a flaw that makes it unsuitable for this sort of test: e.g. starting the playback where the cursor is in the spectrogram a few posts above constitutes a truncation which causes the 20kHz energy at that point to spread over the entire audible spectrum.

As discussed in another thread it seems that foo_abx has a flaw that makes it unsuitable for this sort of test: e.g. starting the playback where the cursor is in the spectrogram a few posts above constitutes a truncation which causes the 20kHz energy at that point to spread over the entire audible spectrum.